Add 'AudioPacket' notification to media transport interface.
So far, base channel was only notifying about 'first audio packet' when
RTP was used, and it never notified about it when media_transport
interface was used. This change adds a sigslot to notify about a new
media packet to the media transport interface.
Bug: webrtc:9719
Change-Id: Ie9230c407f35b1aaa71ba71008ac34ba8869e2d4
Reviewed-on: https://webrtc-review.googlesource.com/c/117249
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26282}diff --git a/api/media_transport_interface.cc b/api/media_transport_interface.cc
index ccdcb91..b5eaa89 100644
--- a/api/media_transport_interface.cc
+++ b/api/media_transport_interface.cc
@@ -157,6 +157,9 @@
return std::unique_ptr<MediaTransportInterface>(nullptr);
}
+MediaTransportInterface::MediaTransportInterface() = default;
+MediaTransportInterface::~MediaTransportInterface() = default;
+
void MediaTransportInterface::SetKeyFrameRequestCallback(
MediaTransportKeyFrameRequestCallback* callback) {}
@@ -168,6 +171,9 @@
void MediaTransportInterface::SetNetworkChangeCallback(
MediaTransportNetworkChangeCallback* callback) {}
+void MediaTransportInterface::SetFirstAudioPacketReceivedObserver(
+ AudioPacketReceivedObserver* observer) {}
+
void MediaTransportInterface::AddTargetTransferRateObserver(
TargetTransferRateObserver* observer) {}
void MediaTransportInterface::RemoveTargetTransferRateObserver(
diff --git a/api/media_transport_interface.h b/api/media_transport_interface.h
index ee1b8e3..a440860 100644
--- a/api/media_transport_interface.h
+++ b/api/media_transport_interface.h
@@ -40,6 +40,15 @@
class RtcEventLog;
+class AudioPacketReceivedObserver {
+ public:
+ virtual ~AudioPacketReceivedObserver() = default;
+
+ // Invoked for the first received audio packet on a given channel id.
+ // It will be invoked once for each channel id.
+ virtual void OnFirstAudioPacketReceived(int64_t channel_id) = 0;
+};
+
// A collection of settings for creation of media transport.
struct MediaTransportSettings final {
MediaTransportSettings();
@@ -352,7 +361,8 @@
// and receiving bandwidth estimate update from congestion control.
class MediaTransportInterface {
public:
- virtual ~MediaTransportInterface() = default;
+ MediaTransportInterface();
+ virtual ~MediaTransportInterface();
// Start asynchronous send of audio frame. The status returned by this method
// only pertains to the synchronous operations (e.g.
@@ -399,6 +409,15 @@
virtual void RemoveTargetTransferRateObserver(
TargetTransferRateObserver* observer);
+ // Sets audio packets observer, which gets informed about incoming audio
+ // packets. Before destruction, the observer must be unregistered by setting
+ // nullptr.
+ //
+ // This method may be temporary, when the multiplexer is implemented (or
+ // multiplexer may use it to demultiplex channel ids).
+ virtual void SetFirstAudioPacketReceivedObserver(
+ AudioPacketReceivedObserver* observer);
+
// Intended for receive side. AddRttObserver registers an observer to be
// called for each RTT measurement, typically once per ACK. Before media
// transport is destructed the observer must be unregistered.
diff --git a/pc/channel.cc b/pc/channel.cc
index 793f1ee..d26a511 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -744,6 +744,9 @@
crypto_options) {}
VoiceChannel::~VoiceChannel() {
+ if (media_transport()) {
+ media_transport()->SetFirstAudioPacketReceivedObserver(nullptr);
+ }
TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
@@ -762,6 +765,19 @@
OnNetworkRouteChanged(absl::make_optional(network_route));
}
+void VoiceChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport) {
+ BaseChannel::Init_w(rtp_transport, media_transport);
+ if (BaseChannel::media_transport()) {
+ this->media_transport()->SetFirstAudioPacketReceivedObserver(this);
+ }
+}
+
+void VoiceChannel::OnFirstAudioPacketReceived(int64_t channel_id) {
+ has_received_packet_ = true;
+ signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FIRSTPACKETRECEIVED);
+}
+
void VoiceChannel::UpdateMediaSendRecvState_w() {
// Render incoming data if we're the active call, and we have the local
// content. We receive data on the default channel and multiplexed streams.
@@ -1034,7 +1050,8 @@
Deinit();
}
-void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
+void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport) {
BaseChannel::Init_w(rtp_transport, /*media_transport=*/nullptr);
media_channel()->SignalDataReceived.connect(this,
&RtpDataChannel::OnDataReceived);
diff --git a/pc/channel.h b/pc/channel.h
index 09836b2..227c593 100644
--- a/pc/channel.h
+++ b/pc/channel.h
@@ -20,6 +20,7 @@
#include "api/call/audio_sink.h"
#include "api/jsep.h"
+#include "api/media_transport_interface.h"
#include "api/rtp_receiver_interface.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
@@ -87,8 +88,8 @@
bool srtp_required,
webrtc::CryptoOptions crypto_options);
virtual ~BaseChannel();
- void Init_w(webrtc::RtpTransportInternal* rtp_transport,
- webrtc::MediaTransportInterface* media_transport);
+ virtual void Init_w(webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport);
// Deinit may be called multiple times and is simply ignored if it's already
// done.
@@ -292,6 +293,8 @@
bool RegisterRtpDemuxerSink();
+ bool has_received_packet_ = false;
+
private:
bool ConnectToRtpTransport();
void DisconnectFromRtpTransport();
@@ -301,6 +304,7 @@
// MediaTransportNetworkChangeCallback override.
void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route) override;
+
rtc::Thread* const worker_thread_;
rtc::Thread* const network_thread_;
rtc::Thread* const signaling_thread_;
@@ -323,7 +327,6 @@
std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
bool writable_ = false;
bool was_ever_writable_ = false;
- bool has_received_packet_ = false;
const bool srtp_required_ = true;
webrtc::CryptoOptions crypto_options_;
@@ -346,7 +349,8 @@
// VoiceChannel is a specialization that adds support for early media, DTMF,
// and input/output level monitoring.
-class VoiceChannel : public BaseChannel {
+class VoiceChannel : public BaseChannel,
+ public webrtc::AudioPacketReceivedObserver {
public:
VoiceChannel(rtc::Thread* worker_thread,
rtc::Thread* network_thread,
@@ -366,6 +370,8 @@
cricket::MediaType media_type() const override {
return cricket::MEDIA_TYPE_AUDIO;
}
+ void Init_w(webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport) override;
private:
// overrides from BaseChannel
@@ -377,6 +383,8 @@
webrtc::SdpType type,
std::string* error_desc) override;
+ void OnFirstAudioPacketReceived(int64_t channel_id) override;
+
// Last AudioSendParameters sent down to the media_channel() via
// SetSendParameters.
AudioSendParameters last_send_params_;
@@ -443,7 +451,9 @@
DtlsTransportInternal* rtcp_dtls_transport,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport);
- void Init_w(webrtc::RtpTransportInternal* rtp_transport);
+ void Init_w(
+ webrtc::RtpTransportInternal* rtp_transport,
+ webrtc::MediaTransportInterface* media_transport = nullptr) override;
virtual bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
